Telecommunication systems are well known in the art, and today's telephone systems employ various multiplexing techniques to transmit telephone signals of many users over a single transmission line, such as wire or fiber-optic cable. Most of these “hard-wired” systems employ a form of Time Division Multiple Access (TDMA).
Typical telephone multiplexing requires sampling of the telephone signal and transmitting the samples at a frequency much higher than the frequency of the telephone signal. To this end, present systems digitally sample and encode the telephone signal, multiplex and transmit the signal, and then receive, demultiplex and decode the signal. One such sampling and encoding system is Pulse Code Modulation (PCM) in which analog voiceband signals are sampled at a rate of 8 kilosamples per second with each sample represented by 8 bits. Consequently, the voiceband signal is converted to a 64 kilobit per second (kb/s) digital signal.
Another form of telecommunication system is the radio telephone system. Radio telephone systems utilize a group of selected radio frequencies (RF) for carrying telephone communication signals between two locations, and typically employ a form of Frequency Division Multiple Access (FDMA). These radio systems, termed wireless communication systems, are used, for example, in rural locations to provide local telephone service or in mobile units to provide mobile communication services.
One category of RF communication systems employs time division multiplexing to allow for TDMA of the FDMA RF communication channels. This method, called FDMA/TDMA and described in U.S. Pat. No. 4,675,863 (incorporated herein by reference), has been employed to increase capacity of RF communication systems. However, RF communication systems are still significantly limited in capacity when compared to hard-wired or fiber-optic communication systems.
Consequently, to increase capacity even further, signal compression techniques have been used to reduce the bandwidth required for transmission of a telephone signal over an RF channel. A typical technique used for voice signals is Residual Linear Predictive Coding (RELP). RELP or similar speech compression algorithms allow a 64 kilobit per second (kb/s) sampled and quantized voice signal to be transmitted over the RF channel as a reduced bit rate (for example, 14.6 kb/s) signal. The receiver reconstructs the 64 kb/s voice signal from the reduced bit rate signal, and the listener perceives little or no loss in signal quality.
The underlying method of speech compression, including RELP, is an encoding and decoding algorithm which assumes certain characteristics of the harmonics of the human voice. Today, however, a large portion of the communication signals within a telephone network are data communications signals such as facsimile (FAX) or voiceband modem data. Unfortunately, RELP algorithms are not particularly compatible with these data communications signals because the data signals do not exhibit the harmonic characteristics of voice signals.
Accordingly, RF communication systems monitor the received signal to detect the presence of a data communication signal. Typically, data signals representing either FAX or voiceband modem data signals up to 2.4 kb/s (low speed data) have been detected and provided a specialized compression algorithm. The receiver reconstructs the data signal without reducing the transmission data rate. Such a system and method is disclosed in, for example, U.S. Pat. No. 4,974,099 (incorporated herein by reference). Today's telephone data signals, however, are more typically 9.6 kb/s (high speed data) or higher (ultra high speed data, such as 14.4 kb/s), and the present compression techniques do not compress these higher data speeds satisfactorily. Compression of these higher data rates, and especially multiple encodings of these higher data rates, cause a degradation of modem or FAX signal quality, and the modem or FAX machine will typically reduce the data transmission rate when the signals are passed through a RF communication system.